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authorMauro Carvalho Chehab <mchehab@redhat.com>2011-12-30 16:59:37 +0100
committerMauro Carvalho Chehab <mchehab@redhat.com>2011-12-30 16:59:37 +0100
commitb4d48c942c17ce3d3a330ad91e109e522bc97378 (patch)
tree3365292f3a5a502edb51492d011fd326c930ca40 /sound/soc
parent[media] tda10021: Add support for DVB-C Annex C (diff)
parentLinux 3.2-rc7 (diff)
downloadlinux-b4d48c942c17ce3d3a330ad91e109e522bc97378.tar.xz
linux-b4d48c942c17ce3d3a330ad91e109e522bc97378.zip
Merge tag 'v3.2-rc7' into staging/for_v3.3
Linux 3.2-rc7 * tag 'v3.2-rc7': (1304 commits) Linux 3.2-rc7 netfilter: xt_connbytes: handle negation correctly Btrfs: call d_instantiate after all ops are setup Btrfs: fix worker lock misuse in find_worker net: relax rcvbuf limits rps: fix insufficient bounds checking in store_rps_dev_flow_table_cnt() net: introduce DST_NOPEER dst flag mqprio: Avoid panic if no options are provided bridge: provide a mtu() method for fake_dst_ops md/bitmap: It is OK to clear bits during recovery. md: don't give up looking for spares on first failure-to-add md/raid5: ensure correct assessment of drives during degraded reshape. md/linear: fix hot-add of devices to linear arrays. sparc64: Fix MSIQ HV call ordering in pci_sun4v_msiq_build_irq(). pata_of_platform: Add missing CONFIG_OF_IRQ dependency. ipv4: using prefetch requires including prefetch.h VFS: Fix race between CPU hotplug and lglocks vfs: __read_cache_page should use gfp argument rather than GFP_KERNEL USB: Fix usb/isp1760 build on sparc net: Add a flow_cache_flush_deferred function ... Conflicts: drivers/media/common/tuners/tda18218.c drivers/media/video/omap3isp/ispccdc.c drivers/staging/media/as102/as102_drv.h
Diffstat (limited to 'sound/soc')
-rw-r--r--sound/soc/atmel/Kconfig21
-rw-r--r--sound/soc/atmel/Makefile4
-rw-r--r--sound/soc/atmel/playpaq_wm8510.c473
-rw-r--r--sound/soc/codecs/Kconfig2
-rw-r--r--sound/soc/codecs/ad1836.h2
-rw-r--r--sound/soc/codecs/adau1373.c2
-rw-r--r--sound/soc/codecs/cs4270.c10
-rw-r--r--sound/soc/codecs/cs4271.c8
-rw-r--r--sound/soc/codecs/cs42l51.c2
-rw-r--r--sound/soc/codecs/jz4740.c1
-rw-r--r--sound/soc/codecs/max9877.c10
-rw-r--r--sound/soc/codecs/rt5631.c2
-rw-r--r--sound/soc/codecs/sgtl5000.c2
-rw-r--r--sound/soc/codecs/sta32x.c63
-rw-r--r--sound/soc/codecs/sta32x.h1
-rw-r--r--sound/soc/codecs/uda1380.c4
-rw-r--r--sound/soc/codecs/wm8731.c1
-rw-r--r--sound/soc/codecs/wm8753.c3
-rw-r--r--sound/soc/codecs/wm8958-dsp2.c2
-rw-r--r--sound/soc/codecs/wm8962.c4
-rw-r--r--sound/soc/codecs/wm8993.c2
-rw-r--r--sound/soc/codecs/wm8994.c19
-rw-r--r--sound/soc/codecs/wm8996.c1
-rw-r--r--sound/soc/codecs/wm9081.c10
-rw-r--r--sound/soc/codecs/wm9090.c6
-rw-r--r--sound/soc/codecs/wm_hubs.c2
-rw-r--r--sound/soc/fsl/fsl_ssi.c1
-rw-r--r--sound/soc/fsl/mpc8610_hpcd.c24
-rw-r--r--sound/soc/imx/Kconfig2
-rw-r--r--sound/soc/kirkwood/Kconfig3
-rw-r--r--sound/soc/mxs/mxs-pcm.c3
-rw-r--r--sound/soc/mxs/mxs-sgtl5000.c1
-rw-r--r--sound/soc/nuc900/nuc900-ac97.c3
-rw-r--r--sound/soc/pxa/Kconfig3
-rw-r--r--sound/soc/pxa/hx4700.c5
-rw-r--r--sound/soc/samsung/jive_wm8750.c3
-rw-r--r--sound/soc/samsung/smdk2443_wm9710.c1
-rw-r--r--sound/soc/samsung/smdk_wm8994.c1
-rw-r--r--sound/soc/samsung/speyside.c2
-rw-r--r--sound/soc/soc-core.c6
-rw-r--r--sound/soc/soc-utils.c31
41 files changed, 187 insertions, 559 deletions
diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig
index bee3c94f58b0..d1fcc816ce97 100644
--- a/sound/soc/atmel/Kconfig
+++ b/sound/soc/atmel/Kconfig
@@ -1,6 +1,6 @@
config SND_ATMEL_SOC
tristate "SoC Audio for the Atmel System-on-Chip"
- depends on ARCH_AT91 || AVR32
+ depends on ARCH_AT91
help
Say Y or M if you want to add support for codecs attached to
the ATMEL SSC interface. You will also need
@@ -24,25 +24,6 @@ config SND_AT91_SOC_SAM9G20_WM8731
Say Y if you want to add support for SoC audio on WM8731-based
AT91sam9g20 evaluation board.
-config SND_AT32_SOC_PLAYPAQ
- tristate "SoC Audio support for PlayPaq with WM8510"
- depends on SND_ATMEL_SOC && BOARD_PLAYPAQ && AT91_PROGRAMMABLE_CLOCKS
- select SND_ATMEL_SOC_SSC
- select SND_SOC_WM8510
- help
- Say Y or M here if you want to add support for SoC audio
- on the LRS PlayPaq.
-
-config SND_AT32_SOC_PLAYPAQ_SLAVE
- bool "Run CODEC on PlayPaq in slave mode"
- depends on SND_AT32_SOC_PLAYPAQ
- default n
- help
- Say Y if you want to run with the AT32 SSC generating the BCLK
- and FRAME signals on the PlayPaq. Unless you want to play
- with the AT32 as the SSC master, you probably want to say N here,
- as this will give you better sound quality.
-
config SND_AT91_SOC_AFEB9260
tristate "SoC Audio support for AFEB9260 board"
depends on ARCH_AT91 && MACH_AFEB9260 && SND_ATMEL_SOC
diff --git a/sound/soc/atmel/Makefile b/sound/soc/atmel/Makefile
index e7ea56bd5f82..a5c0bf19da78 100644
--- a/sound/soc/atmel/Makefile
+++ b/sound/soc/atmel/Makefile
@@ -8,9 +8,5 @@ obj-$(CONFIG_SND_ATMEL_SOC_SSC) += snd-soc-atmel_ssc_dai.o
# AT91 Machine Support
snd-soc-sam9g20-wm8731-objs := sam9g20_wm8731.o
-# AT32 Machine Support
-snd-soc-playpaq-objs := playpaq_wm8510.o
-
obj-$(CONFIG_SND_AT91_SOC_SAM9G20_WM8731) += snd-soc-sam9g20-wm8731.o
-obj-$(CONFIG_SND_AT32_SOC_PLAYPAQ) += snd-soc-playpaq.o
obj-$(CONFIG_SND_AT91_SOC_AFEB9260) += snd-soc-afeb9260.o
diff --git a/sound/soc/atmel/playpaq_wm8510.c b/sound/soc/atmel/playpaq_wm8510.c
deleted file mode 100644
index 73ae99ad4578..000000000000
--- a/sound/soc/atmel/playpaq_wm8510.c
+++ /dev/null
@@ -1,473 +0,0 @@
-/* sound/soc/at32/playpaq_wm8510.c
- * ASoC machine driver for PlayPaq using WM8510 codec
- *
- * Copyright (C) 2008 Long Range Systems
- * Geoffrey Wossum <gwossum@acm.org>
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License version 2 as
- * published by the Free Software Foundation.
- *
- * This code is largely inspired by sound/soc/at91/eti_b1_wm8731.c
- *
- * NOTE: If you don't have the AT32 enhanced portmux configured (which
- * isn't currently in the mainline or Atmel patched kernel), you will
- * need to set the MCLK pin (PA30) to peripheral A in your board initialization
- * code. Something like:
- * at32_select_periph(GPIO_PIN_PA(30), GPIO_PERIPH_A, 0);
- *
- */
-
-/* #define DEBUG */
-
-#include <linux/module.h>
-#include <linux/moduleparam.h>
-#include <linux/kernel.h>
-#include <linux/errno.h>
-#include <linux/clk.h>
-#include <linux/timer.h>
-#include <linux/interrupt.h>
-#include <linux/platform_device.h>
-
-#include <sound/core.h>
-#include <sound/pcm.h>
-#include <sound/pcm_params.h>
-#include <sound/soc.h>
-
-#include <mach/at32ap700x.h>
-#include <mach/portmux.h>
-
-#include "../codecs/wm8510.h"
-#include "atmel-pcm.h"
-#include "atmel_ssc_dai.h"
-
-
-/*-------------------------------------------------------------------------*\
- * constants
-\*-------------------------------------------------------------------------*/
-#define MCLK_PIN GPIO_PIN_PA(30)
-#define MCLK_PERIPH GPIO_PERIPH_A
-
-
-/*-------------------------------------------------------------------------*\
- * data types
-\*-------------------------------------------------------------------------*/
-/* SSC clocking data */
-struct ssc_clock_data {
- /* CMR div */
- unsigned int cmr_div;
-
- /* Frame period (as needed by xCMR.PERIOD) */
- unsigned int period;
-
- /* The SSC clock rate these settings where calculated for */
- unsigned long ssc_rate;
-};
-
-
-/*-------------------------------------------------------------------------*\
- * module data
-\*-------------------------------------------------------------------------*/
-static struct clk *_gclk0;
-static struct clk *_pll0;
-
-#define CODEC_CLK (_gclk0)
-
-
-/*-------------------------------------------------------------------------*\
- * Sound SOC operations
-\*-------------------------------------------------------------------------*/
-#if defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE
-static struct ssc_clock_data playpaq_wm8510_calc_ssc_clock(
- struct snd_pcm_hw_params *params,
- struct snd_soc_dai *cpu_dai)
-{
- struct at32_ssc_info *ssc_p = snd_soc_dai_get_drvdata(cpu_dai);
- struct ssc_device *ssc = ssc_p->ssc;
- struct ssc_clock_data cd;
- unsigned int rate, width_bits, channels;
- unsigned int bitrate, ssc_div;
- unsigned actual_rate;
-
-
- /*
- * Figure out required bitrate
- */
- rate = params_rate(params);
- channels = params_channels(params);
- width_bits = snd_pcm_format_physical_width(params_format(params));
- bitrate = rate * width_bits * channels;
-
-
- /*
- * Figure out required SSC divider and period for required bitrate
- */
- cd.ssc_rate = clk_get_rate(ssc->clk);
- ssc_div = cd.ssc_rate / bitrate;
- cd.cmr_div = ssc_div / 2;
- if (ssc_div & 1) {
- /* round cmr_div up */
- cd.cmr_div++;
- }
- cd.period = width_bits - 1;
-
-
- /*
- * Find actual rate, compare to requested rate
- */
- actual_rate = (cd.ssc_rate / (cd.cmr_div * 2)) / (2 * (cd.period + 1));
- pr_debug("playpaq_wm8510: Request rate = %u, actual rate = %u\n",
- rate, actual_rate);
-
-
- return cd;
-}
-#endif /* CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE */
-
-
-
-static int playpaq_wm8510_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
-{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
- struct at32_ssc_info *ssc_p = snd_soc_dai_get_drvdata(cpu_dai);
- struct ssc_device *ssc = ssc_p->ssc;
- unsigned int pll_out = 0, bclk = 0, mclk_div = 0;
- int ret;
-
-
- /* Due to difficulties with getting the correct clocks from the AT32's
- * PLL0, we're going to let the CODEC be in charge of all the clocks
- */
-#if !defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE
- const unsigned int fmt = (SND_SOC_DAIFMT_I2S |
- SND_SOC_DAIFMT_NB_NF |
- SND_SOC_DAIFMT_CBM_CFM);
-#else
- struct ssc_clock_data cd;
- const unsigned int fmt = (SND_SOC_DAIFMT_I2S |
- SND_SOC_DAIFMT_NB_NF |
- SND_SOC_DAIFMT_CBS_CFS);
-#endif
-
- if (ssc == NULL) {
- pr_warning("playpaq_wm8510_hw_params: ssc is NULL!\n");
- return -EINVAL;
- }
-
-
- /*
- * Figure out PLL and BCLK dividers for WM8510
- */
- switch (params_rate(params)) {
- case 48000:
- pll_out = 24576000;
- mclk_div = WM8510_MCLKDIV_2;
- bclk = WM8510_BCLKDIV_8;
- break;
-
- case 44100:
- pll_out = 22579200;
- mclk_div = WM8510_MCLKDIV_2;
- bclk = WM8510_BCLKDIV_8;
- break;
-
- case 22050:
- pll_out = 22579200;
- mclk_div = WM8510_MCLKDIV_4;
- bclk = WM8510_BCLKDIV_8;
- break;
-
- case 16000:
- pll_out = 24576000;
- mclk_div = WM8510_MCLKDIV_6;
- bclk = WM8510_BCLKDIV_8;
- break;
-
- case 11025:
- pll_out = 22579200;
- mclk_div = WM8510_MCLKDIV_8;
- bclk = WM8510_BCLKDIV_8;
- break;
-
- case 8000:
- pll_out = 24576000;
- mclk_div = WM8510_MCLKDIV_12;
- bclk = WM8510_BCLKDIV_8;
- break;
-
- default:
- pr_warning("playpaq_wm8510: Unsupported sample rate %d\n",
- params_rate(params));
- return -EINVAL;
- }
-
-
- /*
- * set CPU and CODEC DAI configuration
- */
- ret = snd_soc_dai_set_fmt(codec_dai, fmt);
- if (ret < 0) {
- pr_warning("playpaq_wm8510: "
- "Failed to set CODEC DAI format (%d)\n",
- ret);
- return ret;
- }
- ret = snd_soc_dai_set_fmt(cpu_dai, fmt);
- if (ret < 0) {
- pr_warning("playpaq_wm8510: "
- "Failed to set CPU DAI format (%d)\n",
- ret);
- return ret;
- }
-
-
- /*
- * Set CPU clock configuration
- */
-#if defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE
- cd = playpaq_wm8510_calc_ssc_clock(params, cpu_dai);
- pr_debug("playpaq_wm8510: cmr_div = %d, period = %d\n",
- cd.cmr_div, cd.period);
- ret = snd_soc_dai_set_clkdiv(cpu_dai, AT32_SSC_CMR_DIV, cd.cmr_div);
- if (ret < 0) {
- pr_warning("playpaq_wm8510: Failed to set CPU CMR_DIV (%d)\n",
- ret);
- return ret;
- }
- ret = snd_soc_dai_set_clkdiv(cpu_dai, AT32_SSC_TCMR_PERIOD,
- cd.period);
- if (ret < 0) {
- pr_warning("playpaq_wm8510: "
- "Failed to set CPU transmit period (%d)\n",
- ret);
- return ret;
- }
-#endif /* CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE */
-
-
- /*
- * Set CODEC clock configuration
- */
- pr_debug("playpaq_wm8510: "
- "pll_in = %ld, pll_out = %u, bclk = %x, mclk = %x\n",
- clk_get_rate(CODEC_CLK), pll_out, bclk, mclk_div);
-
-
-#if !defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE
- ret = snd_soc_dai_set_clkdiv(codec_dai, WM8510_BCLKDIV, bclk);
- if (ret < 0) {
- pr_warning
- ("playpaq_wm8510: Failed to set CODEC DAI BCLKDIV (%d)\n",
- ret);
- return ret;
- }
-#endif /* CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE */
-
-
- ret = snd_soc_dai_set_pll(codec_dai, 0, 0,
- clk_get_rate(CODEC_CLK), pll_out);
- if (ret < 0) {
- pr_warning("playpaq_wm8510: Failed to set CODEC DAI PLL (%d)\n",
- ret);
- return ret;
- }
-
-
- ret = snd_soc_dai_set_clkdiv(codec_dai, WM8510_MCLKDIV, mclk_div);
- if (ret < 0) {
- pr_warning("playpaq_wm8510: Failed to set CODEC MCLKDIV (%d)\n",
- ret);
- return ret;
- }
-
-
- return 0;
-}
-
-
-
-static struct snd_soc_ops playpaq_wm8510_ops = {
- .hw_params = playpaq_wm8510_hw_params,
-};
-
-
-
-static const struct snd_soc_dapm_widget playpaq_dapm_widgets[] = {
- SND_SOC_DAPM_MIC("Int Mic", NULL),
- SND_SOC_DAPM_SPK("Ext Spk", NULL),
-};
-
-
-
-static const struct snd_soc_dapm_route intercon[] = {
- /* speaker connected to SPKOUT */
- {"Ext Spk", NULL, "SPKOUTP"},
- {"Ext Spk", NULL, "SPKOUTN"},
-
- {"Mic Bias", NULL, "Int Mic"},
- {"MICN", NULL, "Mic Bias"},
- {"MICP", NULL, "Mic Bias"},
-};
-
-
-
-static int playpaq_wm8510_init(struct snd_soc_pcm_runtime *rtd)
-{
- struct snd_soc_codec *codec = rtd->codec;
- struct snd_soc_dapm_context *dapm = &codec->dapm;
- int i;
-
- /*
- * Add DAPM widgets
- */
- for (i = 0; i < ARRAY_SIZE(playpaq_dapm_widgets); i++)
- snd_soc_dapm_new_control(dapm, &playpaq_dapm_widgets[i]);
-
-
-
- /*
- * Setup audio path interconnects
- */
- snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon));
-
-
-
- /* always connected pins */
- snd_soc_dapm_enable_pin(dapm, "Int Mic");
- snd_soc_dapm_enable_pin(dapm, "Ext Spk");
-
-
-
- /* Make CSB show PLL rate */
- snd_soc_dai_set_clkdiv(rtd->codec_dai, WM8510_OPCLKDIV,
- WM8510_OPCLKDIV_1 | 4);
-
- return 0;
-}
-
-
-
-static struct snd_soc_dai_link playpaq_wm8510_dai = {
- .name = "WM8510",
- .stream_name = "WM8510 PCM",
- .cpu_dai_name= "atmel-ssc-dai.0",
- .platform_name = "atmel-pcm-audio",
- .codec_name = "wm8510-codec.0-0x1a",
- .codec_dai_name = "wm8510-hifi",
- .init = playpaq_wm8510_init,
- .ops = &playpaq_wm8510_ops,
-};
-
-
-
-static struct snd_soc_card snd_soc_playpaq = {
- .name = "LRS_PlayPaq_WM8510",
- .dai_link = &playpaq_wm8510_dai,
- .num_links = 1,
-};
-
-static struct platform_device *playpaq_snd_device;
-
-
-static int __init playpaq_asoc_init(void)
-{
- int ret = 0;
-
- /*
- * Configure MCLK for WM8510
- */
- _gclk0 = clk_get(NULL, "gclk0");
- if (IS_ERR(_gclk0)) {
- _gclk0 = NULL;
- ret = PTR_ERR(_gclk0);
- goto err_gclk0;
- }
- _pll0 = clk_get(NULL, "pll0");
- if (IS_ERR(_pll0)) {
- _pll0 = NULL;
- ret = PTR_ERR(_pll0);
- goto err_pll0;
- }
- ret = clk_set_parent(_gclk0, _pll0);
- if (ret) {
- pr_warning("snd-soc-playpaq: "
- "Failed to set PLL0 as parent for DAC clock\n");
- goto err_set_clk;
- }
- clk_set_rate(CODEC_CLK, 12000000);
- clk_enable(CODEC_CLK);
-
-#if defined CONFIG_AT32_ENHANCED_PORTMUX
- at32_select_periph(MCLK_PIN, MCLK_PERIPH, 0);
-#endif
-
-
- /*
- * Create and register platform device
- */
- playpaq_snd_device = platform_device_alloc("soc-audio", 0);
- if (playpaq_snd_device == NULL) {
- ret = -ENOMEM;
- goto err_device_alloc;
- }
-
- platform_set_drvdata(playpaq_snd_device, &snd_soc_playpaq);
-
- ret = platform_device_add(playpaq_snd_device);
- if (ret) {
- pr_warning("playpaq_wm8510: platform_device_add failed (%d)\n",
- ret);
- goto err_device_add;
- }
-
- return 0;
-
-
-err_device_add:
- if (playpaq_snd_device != NULL) {
- platform_device_put(playpaq_snd_device);
- playpaq_snd_device = NULL;
- }
-err_device_alloc:
-err_set_clk:
- if (_pll0 != NULL) {
- clk_put(_pll0);
- _pll0 = NULL;
- }
-err_pll0:
- if (_gclk0 != NULL) {
- clk_put(_gclk0);
- _gclk0 = NULL;
- }
- return ret;
-}
-
-
-static void __exit playpaq_asoc_exit(void)
-{
- if (_gclk0 != NULL) {
- clk_put(_gclk0);
- _gclk0 = NULL;
- }
- if (_pll0 != NULL) {
- clk_put(_pll0);
- _pll0 = NULL;
- }
-
-#if defined CONFIG_AT32_ENHANCED_PORTMUX
- at32_free_pin(MCLK_PIN);
-#endif
-
- platform_device_unregister(playpaq_snd_device);
- playpaq_snd_device = NULL;
-}
-
-module_init(playpaq_asoc_init);
-module_exit(playpaq_asoc_exit);
-
-MODULE_AUTHOR("Geoffrey Wossum <gwossum@acm.org>");
-MODULE_DESCRIPTION("ASoC machine driver for LRS PlayPaq");
-MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 4584514d93d4..fa787d45d74a 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -33,7 +33,7 @@ config SND_SOC_ALL_CODECS
select SND_SOC_CX20442
select SND_SOC_DA7210 if I2C
select SND_SOC_DFBMCS320
- select SND_SOC_JZ4740_CODEC if SOC_JZ4740
+ select SND_SOC_JZ4740_CODEC
select SND_SOC_LM4857 if I2C
select SND_SOC_MAX98088 if I2C
select SND_SOC_MAX98095 if I2C
diff --git a/sound/soc/codecs/ad1836.h b/sound/soc/codecs/ad1836.h
index 444747f0db26..dd7be0dbbc58 100644
--- a/sound/soc/codecs/ad1836.h
+++ b/sound/soc/codecs/ad1836.h
@@ -34,7 +34,7 @@
#define AD1836_ADC_CTRL2 13
#define AD1836_ADC_WORD_LEN_MASK 0x30
-#define AD1836_ADC_WORD_OFFSET 5
+#define AD1836_ADC_WORD_OFFSET 4
#define AD1836_ADC_SERFMT_MASK (7 << 6)
#define AD1836_ADC_SERFMT_PCK256 (0x4 << 6)
#define AD1836_ADC_SERFMT_PCK128 (0x5 << 6)
diff --git a/sound/soc/codecs/adau1373.c b/sound/soc/codecs/adau1373.c
index 1ccf8dd47576..45c63028b40d 100644
--- a/sound/soc/codecs/adau1373.c
+++ b/sound/soc/codecs/adau1373.c
@@ -245,7 +245,7 @@ static const char *adau1373_bass_hpf_cutoff_text[] = {
};
static const unsigned int adau1373_bass_tlv[] = {
- TLV_DB_RANGE_HEAD(4),
+ TLV_DB_RANGE_HEAD(3),
0, 2, TLV_DB_SCALE_ITEM(-600, 600, 1),
3, 4, TLV_DB_SCALE_ITEM(950, 250, 0),
5, 7, TLV_DB_SCALE_ITEM(1400, 150, 0),
diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c
index f1f237ecec2a..73f46eb459f1 100644
--- a/sound/soc/codecs/cs4270.c
+++ b/sound/soc/codecs/cs4270.c
@@ -601,7 +601,6 @@ static int cs4270_soc_suspend(struct snd_soc_codec *codec, pm_message_t mesg)
static int cs4270_soc_resume(struct snd_soc_codec *codec)
{
struct cs4270_private *cs4270 = snd_soc_codec_get_drvdata(codec);
- struct i2c_client *i2c_client = to_i2c_client(codec->dev);
int reg;
regulator_bulk_enable(ARRAY_SIZE(cs4270->supplies),
@@ -612,14 +611,7 @@ static int cs4270_soc_resume(struct snd_soc_codec *codec)
ndelay(500);
/* first restore the entire register cache ... */
- for (reg = CS4270_FIRSTREG; reg <= CS4270_LASTREG; reg++) {
- u8 val = snd_soc_read(codec, reg);
-
- if (i2c_smbus_write_byte_data(i2c_client, reg, val)) {
- dev_err(codec->dev, "i2c write failed\n");
- return -EIO;
- }
- }
+ snd_soc_cache_sync(codec);
/* ... then disable the power-down bits */
reg = snd_soc_read(codec, CS4270_PWRCTL);
diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c
index 23d1bd5dadda..69fde1506fe1 100644
--- a/sound/soc/codecs/cs4271.c
+++ b/sound/soc/codecs/cs4271.c
@@ -434,7 +434,8 @@ static int cs4271_soc_suspend(struct snd_soc_codec *codec, pm_message_t mesg)
{
int ret;
/* Set power-down bit */
- ret = snd_soc_update_bits(codec, CS4271_MODE2, 0, CS4271_MODE2_PDN);
+ ret = snd_soc_update_bits(codec, CS4271_MODE2, CS4271_MODE2_PDN,
+ CS4271_MODE2_PDN);
if (ret < 0)
return ret;
return 0;
@@ -501,8 +502,9 @@ static int cs4271_probe(struct snd_soc_codec *codec)
return ret;
}
- ret = snd_soc_update_bits(codec, CS4271_MODE2, 0,
- CS4271_MODE2_PDN | CS4271_MODE2_CPEN);
+ ret = snd_soc_update_bits(codec, CS4271_MODE2,
+ CS4271_MODE2_PDN | CS4271_MODE2_CPEN,
+ CS4271_MODE2_PDN | CS4271_MODE2_CPEN);
if (ret < 0)
return ret;
ret = snd_soc_update_bits(codec, CS4271_MODE2, CS4271_MODE2_PDN, 0);
diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c
index 8c3c8205d19e..1ee66361f61b 100644
--- a/sound/soc/codecs/cs42l51.c
+++ b/sound/soc/codecs/cs42l51.c
@@ -555,7 +555,7 @@ static int cs42l51_probe(struct snd_soc_codec *codec)
static struct snd_soc_codec_driver soc_codec_device_cs42l51 = {
.probe = cs42l51_probe,
- .reg_cache_size = CS42L51_NUMREGS,
+ .reg_cache_size = CS42L51_NUMREGS + 1,
.reg_word_size = sizeof(u8),
};
diff --git a/sound/soc/codecs/jz4740.c b/sound/soc/codecs/jz4740.c
index e373f8f06907..3e1f4e172bfb 100644
--- a/sound/soc/codecs/jz4740.c
+++ b/sound/soc/codecs/jz4740.c
@@ -15,6 +15,7 @@
#include <linux/module.h>
#include <linux/platform_device.h>
#include <linux/slab.h>
+#include <linux/io.h>
#include <linux/delay.h>
diff --git a/sound/soc/codecs/max9877.c b/sound/soc/codecs/max9877.c
index 9e7e964a5fa3..dcf6f2a1600a 100644
--- a/sound/soc/codecs/max9877.c
+++ b/sound/soc/codecs/max9877.c
@@ -106,13 +106,13 @@ static int max9877_set_2reg(struct snd_kcontrol *kcontrol,
unsigned int mask = mc->max;
unsigned int val = (ucontrol->value.integer.value[0] & mask);
unsigned int val2 = (ucontrol->value.integer.value[1] & mask);
- unsigned int change = 1;
+ unsigned int change = 0;
- if (((max9877_regs[reg] >> shift) & mask) == val)
- change = 0;
+ if (((max9877_regs[reg] >> shift) & mask) != val)
+ change = 1;
- if (((max9877_regs[reg2] >> shift) & mask) == val2)
- change = 0;
+ if (((max9877_regs[reg2] >> shift) & mask) != val2)
+ change = 1;
if (change) {
max9877_regs[reg] &= ~(mask << shift);
diff --git a/sound/soc/codecs/rt5631.c b/sound/soc/codecs/rt5631.c
index 27a078cbb6eb..4646e808b90a 100644
--- a/sound/soc/codecs/rt5631.c
+++ b/sound/soc/codecs/rt5631.c
@@ -177,7 +177,7 @@ static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -95625, 375, 0);
static const DECLARE_TLV_DB_SCALE(in_vol_tlv, -3450, 150, 0);
/* {0, +20, +24, +30, +35, +40, +44, +50, +52}dB */
static unsigned int mic_bst_tlv[] = {
- TLV_DB_RANGE_HEAD(6),
+ TLV_DB_RANGE_HEAD(7),
0, 0, TLV_DB_SCALE_ITEM(0, 0, 0),
1, 1, TLV_DB_SCALE_ITEM(2000, 0, 0),
2, 2, TLV_DB_SCALE_ITEM(2400, 0, 0),
diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c
index d15695d1c273..bbcf921166f7 100644
--- a/sound/soc/codecs/sgtl5000.c
+++ b/sound/soc/codecs/sgtl5000.c
@@ -365,7 +365,7 @@ static const DECLARE_TLV_DB_SCALE(capture_6db_attenuate, -600, 600, 0);
/* tlv for mic gain, 0db 20db 30db 40db */
static const unsigned int mic_gain_tlv[] = {
- TLV_DB_RANGE_HEAD(4),
+ TLV_DB_RANGE_HEAD(2),
0, 0, TLV_DB_SCALE_ITEM(0, 0, 0),
1, 3, TLV_DB_SCALE_ITEM(2000, 1000, 0),
};
diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c
index bb82408ab8e1..d2f37152f940 100644
--- a/sound/soc/codecs/sta32x.c
+++ b/sound/soc/codecs/sta32x.c
@@ -76,6 +76,8 @@ struct sta32x_priv {
unsigned int mclk;
unsigned int format;
+
+ u32 coef_shadow[STA32X_COEF_COUNT];
};
static const DECLARE_TLV_DB_SCALE(mvol_tlv, -12700, 50, 1);
@@ -227,6 +229,7 @@ static int sta32x_coefficient_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec);
int numcoef = kcontrol->private_value >> 16;
int index = kcontrol->private_value & 0xffff;
unsigned int cfud;
@@ -239,6 +242,11 @@ static int sta32x_coefficient_put(struct snd_kcontrol *kcontrol,
snd_soc_write(codec, STA32X_CFUD, cfud);
snd_soc_write(codec, STA32X_CFADDR2, index);
+ for (i = 0; i < numcoef && (index + i < STA32X_COEF_COUNT); i++)
+ sta32x->coef_shadow[index + i] =
+ (ucontrol->value.bytes.data[3 * i] << 16)
+ | (ucontrol->value.bytes.data[3 * i + 1] << 8)
+ | (ucontrol->value.bytes.data[3 * i + 2]);
for (i = 0; i < 3 * numcoef; i++)
snd_soc_write(codec, STA32X_B1CF1 + i,
ucontrol->value.bytes.data[i]);
@@ -252,6 +260,48 @@ static int sta32x_coefficient_put(struct snd_kcontrol *kcontrol,
return 0;
}
+int sta32x_sync_coef_shadow(struct snd_soc_codec *codec)
+{
+ struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec);
+ unsigned int cfud;
+ int i;
+
+ /* preserve reserved bits in STA32X_CFUD */
+ cfud = snd_soc_read(codec, STA32X_CFUD) & 0xf0;
+
+ for (i = 0; i < STA32X_COEF_COUNT; i++) {
+ snd_soc_write(codec, STA32X_CFADDR2, i);
+ snd_soc_write(codec, STA32X_B1CF1,
+ (sta32x->coef_shadow[i] >> 16) & 0xff);
+ snd_soc_write(codec, STA32X_B1CF2,
+ (sta32x->coef_shadow[i] >> 8) & 0xff);
+ snd_soc_write(codec, STA32X_B1CF3,
+ (sta32x->coef_shadow[i]) & 0xff);
+ /* chip documentation does not say if the bits are
+ * self-clearing, so do it explicitly */
+ snd_soc_write(codec, STA32X_CFUD, cfud);
+ snd_soc_write(codec, STA32X_CFUD, cfud | 0x01);
+ }
+ return 0;
+}
+
+int sta32x_cache_sync(struct snd_soc_codec *codec)
+{
+ unsigned int mute;
+ int rc;
+
+ if (!codec->cache_sync)
+ return 0;
+
+ /* mute during register sync */
+ mute = snd_soc_read(codec, STA32X_MMUTE);
+ snd_soc_write(codec, STA32X_MMUTE, mute | STA32X_MMUTE_MMUTE);
+ sta32x_sync_coef_shadow(codec);
+ rc = snd_soc_cache_sync(codec);
+ snd_soc_write(codec, STA32X_MMUTE, mute);
+ return rc;
+}
+
#define SINGLE_COEF(xname, index) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
.info = sta32x_coefficient_info, \
@@ -661,7 +711,7 @@ static int sta32x_set_bias_level(struct snd_soc_codec *codec,
return ret;
}
- snd_soc_cache_sync(codec);
+ sta32x_cache_sync(codec);
}
/* Power up to mute */
@@ -790,6 +840,17 @@ static int sta32x_probe(struct snd_soc_codec *codec)
STA32X_CxCFG_OM_MASK,
2 << STA32X_CxCFG_OM_SHIFT);
+ /* initialize coefficient shadow RAM with reset values */
+ for (i = 4; i <= 49; i += 5)
+ sta32x->coef_shadow[i] = 0x400000;
+ for (i = 50; i <= 54; i++)
+ sta32x->coef_shadow[i] = 0x7fffff;
+ sta32x->coef_shadow[55] = 0x5a9df7;
+ sta32x->coef_shadow[56] = 0x7fffff;
+ sta32x->coef_shadow[59] = 0x7fffff;
+ sta32x->coef_shadow[60] = 0x400000;
+ sta32x->coef_shadow[61] = 0x400000;
+
sta32x_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
/* Bias level configuration will have done an extra enable */
regulator_bulk_disable(ARRAY_SIZE(sta32x->supplies), sta32x->supplies);
diff --git a/sound/soc/codecs/sta32x.h b/sound/soc/codecs/sta32x.h
index b97ee5a75667..d8e32a6262ee 100644
--- a/sound/soc/codecs/sta32x.h
+++ b/sound/soc/codecs/sta32x.h
@@ -19,6 +19,7 @@
/* STA326 register addresses */
#define STA32X_REGISTER_COUNT 0x2d
+#define STA32X_COEF_COUNT 62
#define STA32X_CONFA 0x00
#define STA32X_CONFB 0x01
diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c
index c5ca8cfea60f..0441893e270e 100644
--- a/sound/soc/codecs/uda1380.c
+++ b/sound/soc/codecs/uda1380.c
@@ -863,13 +863,13 @@ static struct i2c_driver uda1380_i2c_driver = {
static int __init uda1380_modinit(void)
{
- int ret;
+ int ret = 0;
#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
ret = i2c_add_driver(&uda1380_i2c_driver);
if (ret != 0)
pr_err("Failed to register UDA1380 I2C driver: %d\n", ret);
#endif
- return 0;
+ return ret;
}
module_init(uda1380_modinit);
diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c
index 7e5ec03f6f8d..a7c9ae17fc7e 100644
--- a/sound/soc/codecs/wm8731.c
+++ b/sound/soc/codecs/wm8731.c
@@ -453,6 +453,7 @@ static int wm8731_set_bias_level(struct snd_soc_codec *codec,
snd_soc_write(codec, WM8731_PWR, 0xffff);
regulator_bulk_disable(ARRAY_SIZE(wm8731->supplies),
wm8731->supplies);
+ codec->cache_sync = 1;
break;
}
codec->dapm.bias_level = level;
diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c
index a9504710bb69..3a629d0d690e 100644
--- a/sound/soc/codecs/wm8753.c
+++ b/sound/soc/codecs/wm8753.c
@@ -190,6 +190,9 @@ static int wm8753_set_dai(struct snd_kcontrol *kcontrol,
struct wm8753_priv *wm8753 = snd_soc_codec_get_drvdata(codec);
u16 ioctl;
+ if (wm8753->dai_func == ucontrol->value.integer.value[0])
+ return 0;
+
if (codec->active)
return -EBUSY;
diff --git a/sound/soc/codecs/wm8958-dsp2.c b/sound/soc/codecs/wm8958-dsp2.c
index 0293763debe5..5a14d5c0e0e1 100644
--- a/sound/soc/codecs/wm8958-dsp2.c
+++ b/sound/soc/codecs/wm8958-dsp2.c
@@ -60,6 +60,8 @@ static int wm8958_dsp2_fw(struct snd_soc_codec *codec, const char *name,
}
if (memcmp(fw->data, "WMFW", 4) != 0) {
+ memcpy(&data32, fw->data, sizeof(data32));
+ data32 = be32_to_cpu(data32);
dev_err(codec->dev, "%s: firmware has bad file magic %08x\n",
name, data32);
goto err;
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index 91d3c6dbeba3..53edd9a8c758 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -1973,7 +1973,7 @@ static int wm8962_reset(struct snd_soc_codec *codec)
static const DECLARE_TLV_DB_SCALE(inpga_tlv, -2325, 75, 0);
static const DECLARE_TLV_DB_SCALE(mixin_tlv, -1500, 300, 0);
static const unsigned int mixinpga_tlv[] = {
- TLV_DB_RANGE_HEAD(7),
+ TLV_DB_RANGE_HEAD(5),
0, 1, TLV_DB_SCALE_ITEM(0, 600, 0),
2, 2, TLV_DB_SCALE_ITEM(1300, 1300, 0),
3, 4, TLV_DB_SCALE_ITEM(1800, 200, 0),
@@ -1988,7 +1988,7 @@ static const DECLARE_TLV_DB_SCALE(bypass_tlv, -1500, 300, 0);
static const DECLARE_TLV_DB_SCALE(out_tlv, -12100, 100, 1);
static const DECLARE_TLV_DB_SCALE(hp_tlv, -700, 100, 0);
static const unsigned int classd_tlv[] = {
- TLV_DB_RANGE_HEAD(7),
+ TLV_DB_RANGE_HEAD(2),
0, 6, TLV_DB_SCALE_ITEM(0, 150, 0),
7, 7, TLV_DB_SCALE_ITEM(1200, 0, 0),
};
diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c
index eec8e1435116..d1a142f48b09 100644
--- a/sound/soc/codecs/wm8993.c
+++ b/sound/soc/codecs/wm8993.c
@@ -512,7 +512,7 @@ static const DECLARE_TLV_DB_SCALE(drc_comp_threash, -4500, 75, 0);
static const DECLARE_TLV_DB_SCALE(drc_comp_amp, -2250, 75, 0);
static const DECLARE_TLV_DB_SCALE(drc_min_tlv, -1800, 600, 0);
static const unsigned int drc_max_tlv[] = {
- TLV_DB_RANGE_HEAD(4),
+ TLV_DB_RANGE_HEAD(2),
0, 2, TLV_DB_SCALE_ITEM(1200, 600, 0),
3, 3, TLV_DB_SCALE_ITEM(3600, 0, 0),
};
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index 9c982e47eb99..d0c545b73d78 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -1325,15 +1325,15 @@ SND_SOC_DAPM_DAC("DAC1R", NULL, WM8994_POWER_MANAGEMENT_5, 0, 0),
};
static const struct snd_soc_dapm_widget wm8994_adc_revd_widgets[] = {
-SND_SOC_DAPM_MUX_E("ADCL Mux", WM8994_POWER_MANAGEMENT_4, 1, 0, &adcl_mux,
- adc_mux_ev, SND_SOC_DAPM_PRE_PMU),
-SND_SOC_DAPM_MUX_E("ADCR Mux", WM8994_POWER_MANAGEMENT_4, 0, 0, &adcr_mux,
- adc_mux_ev, SND_SOC_DAPM_PRE_PMU),
+SND_SOC_DAPM_VIRT_MUX_E("ADCL Mux", WM8994_POWER_MANAGEMENT_4, 1, 0, &adcl_mux,
+ adc_mux_ev, SND_SOC_DAPM_PRE_PMU),
+SND_SOC_DAPM_VIRT_MUX_E("ADCR Mux", WM8994_POWER_MANAGEMENT_4, 0, 0, &adcr_mux,
+ adc_mux_ev, SND_SOC_DAPM_PRE_PMU),
};
static const struct snd_soc_dapm_widget wm8994_adc_widgets[] = {
-SND_SOC_DAPM_MUX("ADCL Mux", WM8994_POWER_MANAGEMENT_4, 1, 0, &adcl_mux),
-SND_SOC_DAPM_MUX("ADCR Mux", WM8994_POWER_MANAGEMENT_4, 0, 0, &adcr_mux),
+SND_SOC_DAPM_VIRT_MUX("ADCL Mux", WM8994_POWER_MANAGEMENT_4, 1, 0, &adcl_mux),
+SND_SOC_DAPM_VIRT_MUX("ADCR Mux", WM8994_POWER_MANAGEMENT_4, 0, 0, &adcr_mux),
};
static const struct snd_soc_dapm_widget wm8994_dapm_widgets[] = {
@@ -2357,6 +2357,11 @@ static int wm8994_hw_params(struct snd_pcm_substream *substream,
bclk |= best << WM8994_AIF1_BCLK_DIV_SHIFT;
lrclk = bclk_rate / params_rate(params);
+ if (!lrclk) {
+ dev_err(dai->dev, "Unable to generate LRCLK from %dHz BCLK\n",
+ bclk_rate);
+ return -EINVAL;
+ }
dev_dbg(dai->dev, "Using LRCLK rate %d for actual LRCLK %dHz\n",
lrclk, bclk_rate / lrclk);
@@ -3178,6 +3183,8 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec)
switch (wm8994->revision) {
case 0:
case 1:
+ case 2:
+ case 3:
wm8994->hubs.dcs_codes_l = -9;
wm8994->hubs.dcs_codes_r = -5;
break;
diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c
index 645c980d6b80..a33b04d17195 100644
--- a/sound/soc/codecs/wm8996.c
+++ b/sound/soc/codecs/wm8996.c
@@ -1968,6 +1968,7 @@ static int wm8996_set_sysclk(struct snd_soc_dai *dai,
break;
case 24576000:
ratediv = WM8996_SYSCLK_DIV;
+ wm8996->sysclk /= 2;
case 12288000:
snd_soc_update_bits(codec, WM8996_AIF_RATE,
WM8996_SYSCLK_RATE, WM8996_SYSCLK_RATE);
diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c
index 3cd35a02c28c..4a398c3bfe84 100644
--- a/sound/soc/codecs/wm9081.c
+++ b/sound/soc/codecs/wm9081.c
@@ -807,7 +807,6 @@ static int wm9081_set_bias_level(struct snd_soc_codec *codec,
mdelay(100);
/* Normal bias enable & soft start off */
- reg |= WM9081_BIAS_ENA;
reg &= ~WM9081_VMID_RAMP;
snd_soc_write(codec, WM9081_VMID_CONTROL, reg);
@@ -818,7 +817,7 @@ static int wm9081_set_bias_level(struct snd_soc_codec *codec,
}
/* VMID 2*240k */
- reg = snd_soc_read(codec, WM9081_BIAS_CONTROL_1);
+ reg = snd_soc_read(codec, WM9081_VMID_CONTROL);
reg &= ~WM9081_VMID_SEL_MASK;
reg |= 0x04;
snd_soc_write(codec, WM9081_VMID_CONTROL, reg);
@@ -830,14 +829,15 @@ static int wm9081_set_bias_level(struct snd_soc_codec *codec,
break;
case SND_SOC_BIAS_OFF:
- /* Startup bias source */
+ /* Startup bias source and disable bias */
reg = snd_soc_read(codec, WM9081_BIAS_CONTROL_1);
reg |= WM9081_BIAS_SRC;
+ reg &= ~WM9081_BIAS_ENA;
snd_soc_write(codec, WM9081_BIAS_CONTROL_1, reg);
- /* Disable VMID and biases with soft ramping */
+ /* Disable VMID with soft ramping */
reg = snd_soc_read(codec, WM9081_VMID_CONTROL);
- reg &= ~(WM9081_VMID_SEL_MASK | WM9081_BIAS_ENA);
+ reg &= ~WM9081_VMID_SEL_MASK;
reg |= WM9081_VMID_RAMP;
snd_soc_write(codec, WM9081_VMID_CONTROL, reg);
diff --git a/sound/soc/codecs/wm9090.c b/sound/soc/codecs/wm9090.c
index 2b5252c9e377..f94c06057c64 100644
--- a/sound/soc/codecs/wm9090.c
+++ b/sound/soc/codecs/wm9090.c
@@ -177,19 +177,19 @@ static void wait_for_dc_servo(struct snd_soc_codec *codec)
}
static const unsigned int in_tlv[] = {
- TLV_DB_RANGE_HEAD(6),
+ TLV_DB_RANGE_HEAD(3),
0, 0, TLV_DB_SCALE_ITEM(-600, 0, 0),
1, 3, TLV_DB_SCALE_ITEM(-350, 350, 0),
4, 6, TLV_DB_SCALE_ITEM(600, 600, 0),
};
static const unsigned int mix_tlv[] = {
- TLV_DB_RANGE_HEAD(4),
+ TLV_DB_RANGE_HEAD(2),
0, 2, TLV_DB_SCALE_ITEM(-1200, 300, 0),
3, 3, TLV_DB_SCALE_ITEM(0, 0, 0),
};
static const DECLARE_TLV_DB_SCALE(out_tlv, -5700, 100, 0);
static const unsigned int spkboost_tlv[] = {
- TLV_DB_RANGE_HEAD(7),
+ TLV_DB_RANGE_HEAD(2),
0, 6, TLV_DB_SCALE_ITEM(0, 150, 0),
7, 7, TLV_DB_SCALE_ITEM(1200, 0, 0),
};
diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c
index 84f33d4ea2cd..48e61e912400 100644
--- a/sound/soc/codecs/wm_hubs.c
+++ b/sound/soc/codecs/wm_hubs.c
@@ -40,7 +40,7 @@ static const DECLARE_TLV_DB_SCALE(outmix_tlv, -2100, 300, 0);
static const DECLARE_TLV_DB_SCALE(spkmixout_tlv, -1800, 600, 1);
static const DECLARE_TLV_DB_SCALE(outpga_tlv, -5700, 100, 0);
static const unsigned int spkboost_tlv[] = {
- TLV_DB_RANGE_HEAD(7),
+ TLV_DB_RANGE_HEAD(2),
0, 6, TLV_DB_SCALE_ITEM(0, 150, 0),
7, 7, TLV_DB_SCALE_ITEM(1200, 0, 0),
};
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index 0268cf989736..83c4bd5b2dd7 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -694,6 +694,7 @@ static int __devinit fsl_ssi_probe(struct platform_device *pdev)
/* Initialize the the device_attribute structure */
dev_attr = &ssi_private->dev_attr;
+ sysfs_attr_init(&dev_attr->attr);
dev_attr->attr.name = "statistics";
dev_attr->attr.mode = S_IRUGO;
dev_attr->show = fsl_sysfs_ssi_show;
diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c
index 31af405bda84..ae49f1c78c6d 100644
--- a/sound/soc/fsl/mpc8610_hpcd.c
+++ b/sound/soc/fsl/mpc8610_hpcd.c
@@ -392,7 +392,8 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev)
}
if (strcasecmp(sprop, "i2s-slave") == 0) {
- machine_data->dai_format = SND_SOC_DAIFMT_I2S;
+ machine_data->dai_format =
+ SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM;
machine_data->codec_clk_direction = SND_SOC_CLOCK_OUT;
machine_data->cpu_clk_direction = SND_SOC_CLOCK_IN;
@@ -409,31 +410,38 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev)
}
machine_data->clk_frequency = be32_to_cpup(iprop);
} else if (strcasecmp(sprop, "i2s-master") == 0) {
- machine_data->dai_format = SND_SOC_DAIFMT_I2S;
+ machine_data->dai_format =
+ SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS;
machine_data->codec_clk_direction = SND_SOC_CLOCK_IN;
machine_data->cpu_clk_direction = SND_SOC_CLOCK_OUT;
} else if (strcasecmp(sprop, "lj-slave") == 0) {
- machine_data->dai_format = SND_SOC_DAIFMT_LEFT_J;
+ machine_data->dai_format =
+ SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_CBM_CFM;
machine_data->codec_clk_direction = SND_SOC_CLOCK_OUT;
machine_data->cpu_clk_direction = SND_SOC_CLOCK_IN;
} else if (strcasecmp(sprop, "lj-master") == 0) {
- machine_data->dai_format = SND_SOC_DAIFMT_LEFT_J;
+ machine_data->dai_format =
+ SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_CBS_CFS;
machine_data->codec_clk_direction = SND_SOC_CLOCK_IN;
machine_data->cpu_clk_direction = SND_SOC_CLOCK_OUT;
} else if (strcasecmp(sprop, "rj-slave") == 0) {
- machine_data->dai_format = SND_SOC_DAIFMT_RIGHT_J;
+ machine_data->dai_format =
+ SND_SOC_DAIFMT_RIGHT_J | SND_SOC_DAIFMT_CBM_CFM;
machine_data->codec_clk_direction = SND_SOC_CLOCK_OUT;
machine_data->cpu_clk_direction = SND_SOC_CLOCK_IN;
} else if (strcasecmp(sprop, "rj-master") == 0) {
- machine_data->dai_format = SND_SOC_DAIFMT_RIGHT_J;
+ machine_data->dai_format =
+ SND_SOC_DAIFMT_RIGHT_J | SND_SOC_DAIFMT_CBS_CFS;
machine_data->codec_clk_direction = SND_SOC_CLOCK_IN;
machine_data->cpu_clk_direction = SND_SOC_CLOCK_OUT;
} else if (strcasecmp(sprop, "ac97-slave") == 0) {
- machine_data->dai_format = SND_SOC_DAIFMT_AC97;
+ machine_data->dai_format =
+ SND_SOC_DAIFMT_AC97 | SND_SOC_DAIFMT_CBM_CFM;
machine_data->codec_clk_direction = SND_SOC_CLOCK_OUT;
machine_data->cpu_clk_direction = SND_SOC_CLOCK_IN;
} else if (strcasecmp(sprop, "ac97-master") == 0) {
- machine_data->dai_format = SND_SOC_DAIFMT_AC97;
+ machine_data->dai_format =
+ SND_SOC_DAIFMT_AC97 | SND_SOC_DAIFMT_CBS_CFS;
machine_data->codec_clk_direction = SND_SOC_CLOCK_IN;
machine_data->cpu_clk_direction = SND_SOC_CLOCK_OUT;
} else {
diff --git a/sound/soc/imx/Kconfig b/sound/soc/imx/Kconfig
index b133bfcc5848..738391757f2c 100644
--- a/sound/soc/imx/Kconfig
+++ b/sound/soc/imx/Kconfig
@@ -28,7 +28,7 @@ config SND_MXC_SOC_WM1133_EV1
config SND_SOC_MX27VIS_AIC32X4
tristate "SoC audio support for Visstrim M10 boards"
- depends on MACH_IMX27_VISSTRIM_M10
+ depends on MACH_IMX27_VISSTRIM_M10 && I2C
select SND_SOC_TLV320AIC32X4
select SND_MXC_SOC_MX2
help
diff --git a/sound/soc/kirkwood/Kconfig b/sound/soc/kirkwood/Kconfig
index 8f49e165f4d1..c62d715235e2 100644
--- a/sound/soc/kirkwood/Kconfig
+++ b/sound/soc/kirkwood/Kconfig
@@ -12,6 +12,7 @@ config SND_KIRKWOOD_SOC_I2S
config SND_KIRKWOOD_SOC_OPENRD
tristate "SoC Audio support for Kirkwood Openrd Client"
depends on SND_KIRKWOOD_SOC && (MACH_OPENRD_CLIENT || MACH_OPENRD_ULTIMATE)
+ depends on I2C
select SND_KIRKWOOD_SOC_I2S
select SND_SOC_CS42L51
help
@@ -20,7 +21,7 @@ config SND_KIRKWOOD_SOC_OPENRD
config SND_KIRKWOOD_SOC_T5325
tristate "SoC Audio support for HP t5325"
- depends on SND_KIRKWOOD_SOC && MACH_T5325
+ depends on SND_KIRKWOOD_SOC && MACH_T5325 && I2C
select SND_KIRKWOOD_SOC_I2S
select SND_SOC_ALC5623
help
diff --git a/sound/soc/mxs/mxs-pcm.c b/sound/soc/mxs/mxs-pcm.c
index dea5aa4aa647..f39d7dd9fbcb 100644
--- a/sound/soc/mxs/mxs-pcm.c
+++ b/sound/soc/mxs/mxs-pcm.c
@@ -357,3 +357,6 @@ static void __exit snd_mxs_pcm_exit(void)
platform_driver_unregister(&mxs_pcm_driver);
}
module_exit(snd_mxs_pcm_exit);
+
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:mxs-pcm-audio");
diff --git a/sound/soc/mxs/mxs-sgtl5000.c b/sound/soc/mxs/mxs-sgtl5000.c
index 7fbeaec06eb4..1c57f6630a48 100644
--- a/sound/soc/mxs/mxs-sgtl5000.c
+++ b/sound/soc/mxs/mxs-sgtl5000.c
@@ -171,3 +171,4 @@ module_exit(mxs_sgtl5000_exit);
MODULE_AUTHOR("Freescale Semiconductor, Inc.");
MODULE_DESCRIPTION("MXS ALSA SoC Machine driver");
MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:mxs-sgtl5000");
diff --git a/sound/soc/nuc900/nuc900-ac97.c b/sound/soc/nuc900/nuc900-ac97.c
index 9c0edad90d8b..a4e3237956e2 100644
--- a/sound/soc/nuc900/nuc900-ac97.c
+++ b/sound/soc/nuc900/nuc900-ac97.c
@@ -365,7 +365,8 @@ static int __devinit nuc900_ac97_drvprobe(struct platform_device *pdev)
if (ret)
goto out3;
- mfp_set_groupg(nuc900_audio->dev); /* enbale ac97 multifunction pin*/
+ /* enbale ac97 multifunction pin */
+ mfp_set_groupg(nuc900_audio->dev, "nuc900-audio");
return 0;
diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig
index ffd2242e305f..a0f7d3cfa470 100644
--- a/sound/soc/pxa/Kconfig
+++ b/sound/soc/pxa/Kconfig
@@ -151,6 +151,7 @@ config SND_SOC_ZYLONITE
config SND_SOC_RAUMFELD
tristate "SoC Audio support Raumfeld audio adapter"
depends on SND_PXA2XX_SOC && (MACH_RAUMFELD_SPEAKER || MACH_RAUMFELD_CONNECTOR)
+ depends on I2C && SPI_MASTER
select SND_PXA_SOC_SSP
select SND_SOC_CS4270
select SND_SOC_AK4104
@@ -159,7 +160,7 @@ config SND_SOC_RAUMFELD
config SND_PXA2XX_SOC_HX4700
tristate "SoC Audio support for HP iPAQ hx4700"
- depends on SND_PXA2XX_SOC && MACH_H4700
+ depends on SND_PXA2XX_SOC && MACH_H4700 && I2C
select SND_PXA2XX_SOC_I2S
select SND_SOC_AK4641
help
diff --git a/sound/soc/pxa/hx4700.c b/sound/soc/pxa/hx4700.c
index 65c124831a00..c664e33fb6d7 100644
--- a/sound/soc/pxa/hx4700.c
+++ b/sound/soc/pxa/hx4700.c
@@ -209,9 +209,10 @@ static int __devinit hx4700_audio_probe(struct platform_device *pdev)
snd_soc_card_hx4700.dev = &pdev->dev;
ret = snd_soc_register_card(&snd_soc_card_hx4700);
if (ret)
- return ret;
+ gpio_free_array(hx4700_audio_gpios,
+ ARRAY_SIZE(hx4700_audio_gpios));
- return 0;
+ return ret;
}
static int __devexit hx4700_audio_remove(struct platform_device *pdev)
diff --git a/sound/soc/samsung/jive_wm8750.c b/sound/soc/samsung/jive_wm8750.c
index 1826acf20f7c..8e523fd9189e 100644
--- a/sound/soc/samsung/jive_wm8750.c
+++ b/sound/soc/samsung/jive_wm8750.c
@@ -101,7 +101,6 @@ static int jive_wm8750_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_codec *codec = rtd->codec;
struct snd_soc_dapm_context *dapm = &codec->dapm;
- int err;
/* These endpoints are not being used. */
snd_soc_dapm_nc_pin(dapm, "LINPUT2");
@@ -131,7 +130,7 @@ static struct snd_soc_card snd_soc_machine_jive = {
.dai_link = &jive_dai,
.num_links = 1,
- .dapm_widgtets = wm8750_dapm_widgets,
+ .dapm_widgets = wm8750_dapm_widgets,
.num_dapm_widgets = ARRAY_SIZE(wm8750_dapm_widgets),
.dapm_routes = audio_map,
.num_dapm_routes = ARRAY_SIZE(audio_map),
diff --git a/sound/soc/samsung/smdk2443_wm9710.c b/sound/soc/samsung/smdk2443_wm9710.c
index 3a0dbfc793f0..8bd1dc5706bf 100644
--- a/sound/soc/samsung/smdk2443_wm9710.c
+++ b/sound/soc/samsung/smdk2443_wm9710.c
@@ -12,6 +12,7 @@
*
*/
+#include <linux/module.h>
#include <sound/soc.h>
static struct snd_soc_card smdk2443;
diff --git a/sound/soc/samsung/smdk_wm8994.c b/sound/soc/samsung/smdk_wm8994.c
index f75e43997d5b..ad9ac42522e2 100644
--- a/sound/soc/samsung/smdk_wm8994.c
+++ b/sound/soc/samsung/smdk_wm8994.c
@@ -9,6 +9,7 @@
#include "../codecs/wm8994.h"
#include <sound/pcm_params.h>
+#include <linux/module.h>
/*
* Default CFG switch settings to use this driver:
diff --git a/sound/soc/samsung/speyside.c b/sound/soc/samsung/speyside.c
index 85bf541a771d..4b8e35410eb1 100644
--- a/sound/soc/samsung/speyside.c
+++ b/sound/soc/samsung/speyside.c
@@ -191,7 +191,7 @@ static int speyside_late_probe(struct snd_soc_card *card)
snd_soc_dapm_ignore_suspend(&card->dapm, "Headset Mic");
snd_soc_dapm_ignore_suspend(&card->dapm, "Main AMIC");
snd_soc_dapm_ignore_suspend(&card->dapm, "Main DMIC");
- snd_soc_dapm_ignore_suspend(&card->dapm, "Speaker");
+ snd_soc_dapm_ignore_suspend(&card->dapm, "Main Speaker");
snd_soc_dapm_ignore_suspend(&card->dapm, "WM1250 Output");
snd_soc_dapm_ignore_suspend(&card->dapm, "WM1250 Input");
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index a5d3685a5d38..a25fa63ce9a2 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -709,6 +709,12 @@ int snd_soc_resume(struct device *dev)
struct snd_soc_card *card = dev_get_drvdata(dev);
int i, ac97_control = 0;
+ /* If the initialization of this soc device failed, there is no codec
+ * associated with it. Just bail out in this case.
+ */
+ if (list_empty(&card->codec_dev_list))
+ return 0;
+
/* AC97 devices might have other drivers hanging off them so
* need to resume immediately. Other drivers don't have that
* problem and may take a substantial amount of time to resume
diff --git a/sound/soc/soc-utils.c b/sound/soc/soc-utils.c
index 0c12b98484bd..4220bb0f2730 100644
--- a/sound/soc/soc-utils.c
+++ b/sound/soc/soc-utils.c
@@ -58,7 +58,36 @@ int snd_soc_params_to_bclk(struct snd_pcm_hw_params *params)
}
EXPORT_SYMBOL_GPL(snd_soc_params_to_bclk);
-static struct snd_soc_platform_driver dummy_platform;
+static const struct snd_pcm_hardware dummy_dma_hardware = {
+ .formats = 0xffffffff,
+ .channels_min = 1,
+ .channels_max = UINT_MAX,
+
+ /* Random values to keep userspace happy when checking constraints */
+ .info = SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER,
+ .buffer_bytes_max = 128*1024,
+ .period_bytes_min = PAGE_SIZE,
+ .period_bytes_max = PAGE_SIZE*2,
+ .periods_min = 2,
+ .periods_max = 128,
+};
+
+static int dummy_dma_open(struct snd_pcm_substream *substream)
+{
+ snd_soc_set_runtime_hwparams(substream, &dummy_dma_hardware);
+
+ return 0;
+}
+
+static struct snd_pcm_ops dummy_dma_ops = {
+ .open = dummy_dma_open,
+ .ioctl = snd_pcm_lib_ioctl,
+};
+
+static struct snd_soc_platform_driver dummy_platform = {
+ .ops = &dummy_dma_ops,
+};
static __devinit int snd_soc_dummy_probe(struct platform_device *pdev)
{